To me sounds like an electrical interference. I assume you're not using
PoE. Try plugging the phone to someplace else and see if the noise
still appears.
Use wifi and battery powered cellphones with SIP clients to rule out
- pbx
- parts of the wired network
To confirm it's an electrical
My approach (in theory only, so please correct me if I'm wrong) would be
to run asterisk on multiple boxes (one each). A dedicated monitoring
box (nagios? custom scripts?) would perform frequent checks against the
boxes (one of my previous projects one asterisk was using call files to
Hi,
On 11/1/2013 5:02 PM, A J Stiles wrote:
You probably will have the most success with Android, because you are going to
need well-documented Source Code to stand a chance of getting anywhere. You
will need an Open Source SIP client and the Source Code for the stock Android
GSM telephony
but i do not know how to interface the CDRs.
has anyone used this tool or any other similar tool?
how about something like this:
pbx@pbx:~$ grep -v ^; /etc/asterisk/cdr_mysql.conf
[global]
hostname=localhost
dbname=dbname
table=tablename
password=password
user=username
port=3306
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign to double zeros if your provider can't handle it
; normal 00 prefix
exten = _00ZZXXX.,1,Macro(beforealldials)
exten =
On 3/21/2013 6:58 PM, Optical Phoenix wrote:
the sip string should be Call-Info:\;answer-after=0. I have not tested
this yet however.
I can confirm this,
exten = 1234,n,SIPAddHeader(Call-Info:\;answer-after=0)
is working with a SPA504 phone.
regards
adam
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Hello Hans,
On 11-16-2011 14:46, Hans Goossen wrote:
I guess some billing solution can do the trick, but I think it's too much for
that little. I don't need any other feature.
i would create a macro which calls an agi. The agi searches the CDR
table (mine is in sql) and calculates if the
On 03-31-2011 01:10, bilal ghayyad wrote:
I can not do the configuration from the web based of the Phone?
it depends on the model. 7912 has a web ui, 79[46]0 does not.
* Can I understand that working with Asterisk does not give a chance to have
IP Phones with featues assigned on the
Good morning,
from the last question i assume you're looking for a SIP-based
configureation.
On 03-30-2011 00:16, bilal ghayyad wrote:
1) How I can assign for each button an extension?
you can configure them as lines (at least in my 7940). look for
linex_name, linex_authname and
On 03-29-2011 19:25, Steve Edwards wrote:
Really? How many callers are you expecting from North Korea, Libya, China,
Iran, etc?
after reviewing last week's log i'd say around 25-28k/min :)
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Hi Kyle,
On 01-20-2011 20:41, Kyle Kienapfel wrote:
I understood that option worked the other way around so attacker
thinks peer name is invalid even when they hit a real one.
sorry, it must be because i'm not a native english speaker but i don't
exactly get what you mean by the above.
to
Postal mail... heh... nice :)
On 12-13-2010 15:49, Thomas Perron wrote:
How do I set up an Exchange or other Mail MX server to interoperate
with VoiceMail?
Not sure if this is an asterisk issue at all. After setting the trivial
options in voicemail.conf, it's really just SMTP and relaying
Hi guys,
i've seen this too, nagios woke me up because it was an extremely high
volume of tries.
I took a peek into the logs and saw that the attacker's script was
trying extensions from 1 to and then random names. I can see the
log in the messages file that several attempts failed
Hi,
On 08-02-2010 20:55, Gordon Henderson wrote:
I generated invoices with PHP code - it uses a LaTeX template which it
fills in the gaps, then feeds it through LaTeX and dvi2pdf to generate
PDFs.
Bit of a geek solution though.
well, then i must be a geek too, because i also decided to
Good morning,
i've noticed many times that there are IVRs that play a ring tone just
before bridging me to an agent. My asterisk does not behave like this
but i've always wanted to.
I'm now playing with 1.6.2.9 and i've read in queue's doc:
On 05-05-2010 18:00, Jian Gao wrote:
In my system (Asterisk 1.4.30) I found that if I have some playback() or
saydigit() before dial(), the billsec in CDR count all the time includes
the playback time. For example, if I dial a number, listen the playback,
then just hangup before the call get
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID 713xxx is dialed send an email to mmos...@xxx.com. with the
time date and CID included in the email. I know how to code some but am
looking for the best way to do this.
something like this?
exten =
Hi,
On 05-04-2010 18:46, isca...@free.fr wrote:
- Create a VPN using OpenVPN
= impossible for me , i'm not admin of the Windows system.
this is a bad thing, but the vpn concept might work after all. have you
considered a pptp/l2tp/ipsec vpn? AFAIK on the client side, you may
succeed
Hi List,
i have to put an * between two other SIP gateways and due to some
circumstances, i have to use sip over tcp. With 1.6.2.6 this is working
fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B
(ocs) and that's about it. In the other direction however (ocs - me -
Hi Guys,
On 04-23-2010 21:40, Nathan Clemons wrote:
SIP is just the control protocol, and can be negotiated over TCP or UDP. The
actual payload is done over RTP, which is a UDP-based protocol.
thanks, for both of you for pointing this out. i was obviously on the
wrong track here. since i
To get back to the original poster's possible situation, i've seen this
with my first IP phone, which was a cisco 7912 (SIP image). With that
phone, asterisk sometimes gave me this same error. I'm quite sure i've
asked the very same question here back then (probably i was a bit more
specific
Is there any program Asterisk users use to calculate ASR and ACD ??
i calculate them from CDR, using php in a fancy webpage. But now as i think
of it, probably an sql query would do just fine.
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